similar to: Voice mail distribution - missing messages

I have asterisk 1.8.0 installed and I am not able to forward a voicemail from one users mailbox to another user. I have the user log into their mailbox press 8 to forward a message enter the extension of the user I wish to forward too I don't prepend a audio message and press # to send the message to the other user from a debug perspective I don't see any errors. The only message I see

Has anyone seen this (version 1.2)? The following dialplan should result in the voicemail message being delivered two both mailboxes (5555557426@context_name and 5555557427@context_name); exten => 0,1,SetCIDName(OPER ${CALLERIDNAME}) exten => 0,2,Dial(SIP/5555557423&SIP/5555557424&SIP/5555557425&SIP/5555557426&SIP

Hi, We're experiencing a problem with basic VM distribution groups where messages won't be delivered. VM is called with a command like: Voicemail(20&30&39&... CLI Shows: Oct 1 20:54:33 NOTICE[26943]: app_voicemail.c:1990 copy_message: Copying message from 60@default to 50@default Oct 1 20:54:38 WARNING[26943]: app.c:1125 ast_lock_path: Failed to lock path

Hi, Would anyone have a clue on this issue? I'm running asterisk 1.4.13. Trying to get a WAV voicemail file attachment sent to my email address. Voicemail is working fine. Email notification of a new message works fine. However, when I set up voicemail.conf to have an attachment of the file sent to email I get an error message like this on the console: [Feb 17 13:33:37]

Hi, I'm running two boxes side by side, identical specs and setup but with differing dialplans. Both are on ast/zap/libpri versions 1.0.9. Both boxes share the same folder for voicemail, exported via NFS from another file server. Everything was working fine for an extended period of time, until just recently when someone rebooted Box A. Now when I dial an extension associated with a SIP

Hello, all. I'm running Asterisk 1.6.1.6 on CentOS 5.3 in a multi-tenant environment with IMAP voice mail storage on Zimbra. One of our clients is having a problem when transferring voice mails from one mailbox to another (option 8 in the standard voice application menu) using their Snom 320 and 360 phones. The end results is the final recipient cannot listen to the voicemail. We also email

Hi All, I have an issue on voice mail with odbc in asterisk 11.7 box. Voice message can be received through Google mail but it doesn't show in phone. The error messages is as follow and let me get your kind advice. -- <SIP/0015-00000007> Playing 'auth-thankyou.g722' (language 'en') [2014-05-28 14:55:13] DEBUG[12260][C-00000006]: app_voicemail.c:3824 last_message_index:

I tried to share my spool directory so I could get monitored calls, and now this error comes up when I try to leave a message in any of my voicemail boxes. Mar 24 12:48:35 WARNING[344081]: app_voicemail.c:1488 leave_voicemail: Error opening text file for o utput -- Recording the message Mar 24 12:48:35 WARNING[344081]: file.c:906 ast_writefile: Unable to open file /var/spool/asterisk/v

Hi All, I have an issue with the ODBC voicemail storage option with asterisk. All appears to work fine, however, I get several sql execute warnings. I was wondering if anyone out there could help me get to the bottom of what is causing this and how I could possibly go about rectifying it. The warning message we are getting is as follows: WARNING[30115]: app_voicemail.c:1280 delete_file: SQL

Hi, we're running Asterisk 1.2.10 and have been randomly being left blank voicemails with long messages that we can't hear. I've searched and searched but cannot find a solution. This is what happens: Internal Server runs Asterisk 1.2.10 where our mailboxes are Incoming Server (behind a firewall) runs Asterisk 1.2.13 and calls are bridged between this server and our internal server.

Hello, I have a default installation of asterisk 1.6.1.9-2 When i create a user in users.conf via asterisk-gui, calls, voicemail etc works. But if i create a user realtime (and my realtime caching is available too) i can see the realtime user with sip show peers. But, my local dial rules does not work. I can call from realtime user to static users(the ones in users.conf) and if they are not

I'm running the a very recent CVS version of asterisk on an RH9 machine. My problem is that my x100p takes about 10 seconds to detect a hangup. After that it takes about 10 more seconds for the the zaptel device to release the line. Here's an example of my console report: == Parsing '/var/spool/asterisk/voicemail/default/101/INBOX/msg0000.txt': == Parsing

SOFTWARE FreePBX 2.1.3 CentOS 4.4 Asterisk 1.2.13 Zaptel 1.2.10 Sangoma Wanpipe 2.3.4.5 I had an Asterisk server lock up on me today after 95 days of up time. Had to manually kill the Asterisk process and then restart. Nothing out of the ordinary in terms of memory use as far as I could tell. Seems to be running fine now. Here is the log file. I deleted the stuff in the middle to keep it

Hi, How do I disabled asterisk to use database and storage voicemail in directory. Im getting the below error [Dec 3 19:08:53] WARNING[4934]: app_voicemail.c:3430 inboxcount: Failed to obtain database object for 'asterisk'! [Dec 3 19:08:55] WARNING[5129]: app_voicemail.c:2353 last_message_index: Failed to obtain database object for 'asterisk'! [Dec 3 19:09:04]

Hello, I'm using asterisk-1.0.9 and wanted to compile it with database support in voicemail (postgres). I made the following changes in the /apps/Makefile: USE_MYSQL_VM_INTERFACE=0 USE_POSTGRES_VM_INTERFACE=1 Without theese changes asterisk compiles perfectly, but after setting the postgres variable on 1 (that's what I have to do or not?), asterisk terminates with the following compile

Hello. Has anyone run into problems accessing voicemail with the Sipura SPA-2002's? Our SPA-2000's work fine (registers fine, able to make and receive calls properly & also able to access voicemail). We've configured the 2002's exactly the same way. However, with the SPA-2002 we're unable to access the voicemail system (though it does register fine and is able to

The Asterisk Development Team is pleased to announce the release of Asterisk 1.4.0-beta4. This release contains a number of updates: - a bug fix for the ExternalIVR application and addition of 'silence' sound files to support it - various SIP interoperability improvements - memory and dialog leaks in the SIP channel driver - a fix to music-on-hold random mode that was not really random -

The Asterisk Development Team is pleased to announce the release of Asterisk 1.4.0-beta4. This release contains a number of updates: - a bug fix for the ExternalIVR application and addition of 'silence' sound files to support it - various SIP interoperability improvements - memory and dialog leaks in the SIP channel driver - a fix to music-on-hold random mode that was not really random -

Hello, On a Debian Buster instance, I compiled Asterisk 17.3.0 from source. I enables all 3 File, IMAP and ODBC voicemail modules but I'm still using classical File module (in modules;conf and voicemail.conf): cd asterisk-17.3.0 ... make menuselect.makeopts menuselect/menuselect --enable app_voicemail_imap menuselect.makeopts; done menuselect/menuselect --enable app_voicemail_odbc

Hello steve. Here is a patch I wrote for app_voicemail.c which does exactly as you describe. When the outgoing message is playing, if the listener hits the "*" key, they're prompted for a mailbox and password, whereupon they can check their voicemail as if they were using the internal phone. I found no other way of doing this. If you patch your app_voicemail.c, I have V1.44 from